ESS Sabre Reference DAC (8-channel)

Ivan Petrov said:
I just have to ask the question - in what time frame can we expect to have the kit, I hope I have not gone insane by then? And a three or four figure price? I know, I know, I need to be patient but man, is that one beautiful looking DAC...


Thank you very much for the kind words.

It will be ready for the public very soon. It will not be 4 figures I promise that. :)

I am just tweaking the firmware to make it more configurable and documenting it.

Brian is pretty close to being setup for production runs.

As usual people can get the bare PCBs from us too.

Cheers!
Russ
 
Hi Russ White,
I see that ESS chip integrated volume control function. That is mean I can use this DAC as preamp and I can increase/decrease ouput signal when I play digital source through SPDIF such as CD or SACD? I think it is better than the signal pass preamp separate. Of course, I can't use this function with analog source such as LP...
Do you activate this function into your Buffalo DAC?
Thank
 
langtuhanoi said:
Hi Russ White,
I see that ESS chip integrated volume control function. That is mean I can use this DAC as preamp and I can increase/decrease ouput signal when I play digital source through SPDIF such as CD or SACD? I think it is better than the signal pass preamp separate. Of course, I can't use this function with analog source such as LP...
Do you activate this function into your Buffalo DAC?
Thank


The DAC supports I2C input, I have already rigged a simple external controller to adjust volume (a JT controller rigged for the purpose). This way no preamp is required at all as you say, and I do think this is the best way to go.

The Buffalo DAC board itself does not support the volume feature, but we will have a controller module coming very soon which can be easily integrated. Also its really simple to rig one of your own. You could actually easily make the buffalo board do it on its own by choosing a PIC with an ADC like 12F675. That ADC could be fed with a pot between VDD and GND. Then just write the firmware to poll it and set the volume. :)

Lots of DIY options. ;)

Cheers!
Russ
 
fierce_freak said:
Is there any theoretical (or heard) advantage to turning SPDIF into I2S before feeding the Sabre DAC, or should just running the Coax straight into the Sabre yield roughly the same result? It seems they should be comparable from my admittedly newbie-ish perspective.


Will not make any difference. Virtually all jitter is rejected by the ASRC.
 
Re: Re: DAC4392 + ESS Sabre: this is incredible.

For hirez:

I can relate to not quite having words to describe sound that is RFG, really good.

Some words that might make sense: there is good hi-fi, and there is that transcendent experience of "ongaku", in which one does not so just hear, but one senses.

Clocks:

Don't have specific experience with Tent, am happy with my own, which is a differential fet (sometimes cascoded) colpitts with clean power/ground/layout, padded with sorbothane. I plan of trying the Crystek modules for 40 MHz, and if they work as good or better, then why not?

Anybody here sifted through the hype and quantified the phase noise of a Rubidium clock? Found one at 40 MHz that is somewhat affordable? Maybe surplus?

Output stage:

My current (pun!) favorite is a lowish value i/v resistor feeding a Lundahl Amorphous core transformer. Only works for DAC chips that have sufficient output voltage compliance, like the Sabre. Somewhat lowish output voltage, but very liquid, coherent, clean. Capable of ongaku.

I have been on both sides of the fence about transformers: many do indeed sound like the negatives attributed to them: lumpy, loose, steely, granular, lifeless, constricted, diffuse, fuzzy.

However, the Lundahl Amorphous core ones, like the LL1674 as an i/v are entirely another story. Relaxed ease, open, liquid, tight and defines low end, space and imaging galore, details without edge. I like! Look forward to fine tuning with the Sabre.

Cheers,

WMS
 
Cappy said:
Russ,

Thanks for the link.

So "the volume control is done just before the oversampling filter on the digital data."

This sounds more like the input than the output.

The volume is done in the digital domain -- any idea what the bit resolution is?

This is what Dustin said:
This is how the datapath is in the Sabre DAC. 24 bits in, 24 bits coeff, 56bit MAC, 28 bits to the modulator

So I am guessing the attenuation is done at 56bit MAC.

Will let Dustin clarify that if I am off base.
 
fierce_freak said:
Is there any theoretical (or heard) advantage to turning SPDIF into I2S before feeding the Sabre DAC, or should just running the Coax straight into the Sabre yield roughly the same result? It seems they should be comparable from my admittedly newbie-ish perspective.

I tried both and didn't notice any sound difference.

The I2S will allow you to use a slower master clock. A 37M clock will be good enough for 24/192 using I2S but you will need a 74M clock for SPDIF. If you don't plan on listening to 192K, then it won't matter much.
 
Re: Re: Re: DAC4392 + ESS Sabre: this is incredible.

wildmonkeysects said:


Anybody here sifted through the hype and quantified the phase noise of a Rubidium clock? Found one at 40 MHz that is somewhat affordable? Maybe surplus?


I looked at Rubidium and they didn't have low phase noise. They are an extremely accurate timebase, but that isn't worth the expense in a digital audio application.

I have a quote from a supplier that makes oscillators with ridiculous low phase noise for the military. He wanted $200 each for a minimum of 5 oscillators and it would be 10 weeks delivery. Those were 50MHz. ouch!

At least they were 3.3V and CMOS output levels. A lot of the expensive low phase noise clocks that I looked at were higher power and had low level sinewave outputs.
 
Russ White said:


This is what Dustin said:


So I am guessing the attenuation is done at 56bit MAC.

Will let Dustin clarify that if I am off base.



Volume in this DAC is done just before the data goes into the FIR filter. Its not just a multiply by a scale factor type volume control. Its a circuit that Martin Mallinson first patented, then I added onto in a further patent, so he is the one to be credited with the idea. It uses a scheme that forces a logarithmic repsonce from one level to the next. The algorithm uses 28 bits to keep the roudning errors minimised. On top of that, when the voluem moves to a new level, it moves in steps f 1/64th of a dB, this is to make sure the "clicking" that can be heard on some implematations is not audible.


As an aside, the ASRC patent in no longer pending, it was issued Feb 12

http://patft.uspto.gov/netacgi/nph-...50&s1=7330138.PN.&OS=PN/7330138&RS=PN/7330138



Thanks

Dustin
 
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DSD data from PCM1791

Hi Russ,

Interesting with the information on how to get DSD data from a player using the PCM1791 D/A.

I checked the data sheet for the 1791 and pin 1 is LRCK, pin 3 is DATA and pin 5 is SCK (dash below). Just to make sure - are those the pins to be used?

Regards,

Jesper
 

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Re: DVD-2910 mod

Russ White said:
Since I have received a few emails asking how its done here is how you get DSD from any SACD player which uses the PCM1791 like the Denon DVD-2910.

Pin 1 is DATA R

Pin 3 is DATA L

Pin 5 is the bit clock.

Then you just need GND.

Cheers!
Russ

As you can see here Russ has said which pins to use. Dustin also mentioned it earlier in the thread. Unless Russ's suggestions were incorrect :) But just pointing out.